1. Field of the Invention
The present invention relates to a communication-status notification apparatus used in gateway equipment of a communication system, and a communication-status display apparatus used in a subscriber terminal connected to the equipment. The invention relates also to a communication-status notification method carried out in the communication-status notification apparatus, a medium in which a communication status notification program used in the equipment is recorded, and a communication apparatus having a communication-status notification mechanism.
2. Description of the Related Art
In recent years, data traffic on various networks (e.g., office networks including telecommunication networks and data-communication networks) has increased constantly. Although being relatively small compared with voice traffic (or audio traffic) in the past, the data traffic accounts for nearly 70 percent of the whole traffic on the present networks, and is expected to continue increasing in the years ahead.
Such a situation has formed the background to produce the concept of transmitting voice data (or audio data) via data-communication networks, thereby unify various kinds of networks into the data-communication networks so as to reduce costs. A technology that is expected to embody this concept is Voice over Internet Protocol (VoIP).
VoIP is a technology of voice-data transmission using an Internet Protocol (IP) network, which is based on IP being a protocol for data transmission. More specifically, VoIP enables to convert voice data into one or more IP packets so as to transmit the IP packets via an IP network, by means of equipment having a mechanism of converting various voice signals such as a telephone signal or a facsimile signal into IP packets, and vice versa (hereinafter called a VoIP-gateway mechanism).
FIG. 22 illustrates a conventional communication system using the VoIP (hereinafter called a VoIP communication system). In the VoIP communication system shown in FIG. 22, a VoIP gateway apparatus 1-1, a VoIP trunk 1-2, an Internet-telephony terminal 1-3 and an Internet-facsimile terminal 1-4 are connected to an IP network 9.
The VoIP gateway apparatus 1-1 is coupled to a public switched telephone network (PSTN) or an integrated services digital network (ISDN) to convert an analog or digital voice signal received from the PSTN or the ISDN into one or more IP packets, and vice versa. The VoIP trunk 1-2 is coupled to an exchange such as a private branch exchange (PBX) to convert a voice signal of the exchange into one or more IP packets, and vice versa.
Receiving various kinds of voice signals (e.g., a telephone signal, which was sent by a telephone, or a facsimile signal, which was sent by a facsimile machine) from the PSTN or the ISDN or via the exchange, the VoIP gateway apparatus 1-1 or the VoIP trunk 1-2 changes the voice signal into voice data so as to create one or more IP packets containing the voice data, and then sends the created IP packets into the IP network 9. And receiving one or more IP packets containing voice data from the IP network 9, the VoIP gateway apparatus 1-1 or the VoIP trunk 1-2 analyzes the received IP packets to extract the contained voice data so as to change the voice data into a voice signal, and then sends the voice signal into the PSTN or the ISDN, or to the exchange.
In short, each of the VoIP gateway apparatus 1-1 and the VoIP trunk 1-2 is provided with the VoIP-gateway mechanism to thereby serve as an interface between a terminal, such as a conventional telephone or a conventional facsimile machine, of an individual subscriber and the IP network 9. The VoIP gateway apparatus 1-1 and the VoIP trunk 1-2 will be generically called VoIP gateway equipment in the following description.
The Internet-telephony terminal 1-3 and the Internet-facsimile terminal 1-4 are directly connected to the IP network 9 so as to serve as a subscriber terminal such as a telephone or a facsimile machine, and also have a function of converting voice signal such as a telephone signal or a facsimile signal into one or more IP packets, and vice versa.
Inputted various voice signals such as a telephone signal or a facsimile signal, the Internet-telephony terminal 1-3 or the Internet-facsimile terminal 1-4 converts the voice signal into voice data to create one or more IP packets containing the voice data, and then sends the created IP packets into the IP network 9. And receiving one or more IP packets containing voice data from the IP network 9, the Internet-telephony terminal 1-3 or the Internet-facsimile terminal 1-4 analyzes the received IP packets to extract the contained voice data, so as to change the extracted voice data into an optimum voice signal such as a telephone signal or a facsimile signal, and then outputs the resultant voice signal.
Each of the Internet-telephony terminal 1-3 and Internet-facsimile terminal 1-4 can be considered as a conventional subscriber terminal, such as a telephone or a facsimile machine, to which the above-mentioned VoIP-gateway mechanism is added. Therefore, in the following description, each of the Internet-telephony terminal 1-3 and the Internet-facsimile terminal 1-4 will be considered as a combination of a mechanism of VoIP gateway equipment and a mechanism of a subscriber terminal formed as a single apparatus, and will be generically called a VoIP gateway terminal.
In short, VoIP is a technology in which VoIP gateway equipment is disposed between a subscriber terminal, such as a telephone or a facsimile machine, and an IP network, thereby enabling to transmit voice data contained in various kinds of voice signals, such as a telephone signal or a facsimile signal, via the IP network.
By putting VoIP to practical use, it is possible to transmit voice data via data-communication networks such as IP networks, thereby unify various kinds of networks into the data-communication networks so as to reduce costs. Further, concurrent transmission of conventional communication data and voice data carries an additional advantage of realizing various new services in which the communication data and the voice data are combined together.
Compared with the conventional technologies of transmitting a voice signal via a PSTN or the like, however, VoIP technology of transmitting voice data via an IP network has also its disadvantages in voice communication on the following points:
{circle around (1)} An individual subscriber is not provided with a dedicated connection line in the IP network. Each IP packet is transmitted across the IP network according to a destination address included in a header of each IP packet. A single link in the IP network would therefore allow various IP packets, directed to different destinations, to pass through.
{circle around (2)} Traffic on the IP network is considerably affected by a kind of application program that each subscriber uses for communication and also by a number of subscribers concurrently connecting to the IP network for communication, and is therefore highly variable according to a communication condition and a period of time in which each subscriber connects to the IP network.
{circle around (3)} Voice-data transmission via the IP network involves creation/analysis process of IP packets (IP-packetization/IP-depacketization process), which increases a transmission delay of voice data.
{circle around (4)} Plural IP packets being in a single connection and having the same destination may take different routs in the IP network, owing to dynamic routing by routers in the IP network. Transmission delays of IP packets even in the same connection therefore vary for every IP packet.
Because of the above-described points, VoIP technology has some disadvantages in QoS (Quality of Service), compared with the conventional voice communication technologies.
Specifically the points {circle around (1)} and {circle around (2)} result in degradation of voice data during the voice communication. Heavy traffic tends to bring about a situation that too many IP packets exceeding the processing capacity of routers are directed to pass through the individual links at once. This situation causes congestion of IP packets, which results in disappearance of any IP packets, and overflow of buffers of the individual routers, which results in abandonment of IP packets subsequently arriving at the individual routers. Such disappearance or abandonment of IP packets can be fully recovered during the ordinal data communication simply by sending the disappeared or abandoned IP packets again, while it affects considerably the voice-data communication, in which immediate or simultaneous communication of voice data is required, because it brings about the degradation of voice data.
The points {circle around (3)} and {circle around (4)} result in delay of voice data during the voice communication. The delay of voice data is increased if the voice data undergoes audio CODEC (code/decode or compression/decompression) processing in the VoIP gateway equipment. Further, because transmission routes of IP packets are unidentifiable in the IP network, it is difficult to accurately estimate the delay of voice data. Such indefinite delay of voice data brings about considerable problem for the voice-data communication, in which immediate or simultaneous communication of voice data is required.
The communication via the IP network also brings about another problem. The Internet, being representative of various IP networks, is composed of a number of interconnected networks and terminals throughout the world. Connecting to the Internet therefore means connecting with the number of networks and terminals throughout the world via the Internet. It is therefore necessary to take adequate security measures for safe connection to the Internet. When a company's intranet is established so as to be connected to the Internet, for example, any security measures such as a firewall or a cryptographic processing would be taken ordinarily. Likewise, when voice VoIP communication system via the Internet is established, appropriate security measures are indispensable in order to prevent tapping or interception of voice data and to thereby ensure the security of voice communication.
Consequently, it is necessary to introduce various measures in respect with communication quality, such as QoS or security, in order to provide a satisfactory VoIP service. Several measures are proposed and actually adopted for ensuring communication quality. As a measure to ensure QoS, for example, it is proposed to apply UDP (user datagram protocol) or RTP (real-time transport protocol), instead of TCP (transmission control protocol) commonly used, for use in combination with IP, or to apply RSVP (resource reservation setup protocol) to a layer higher than that of TCP/IP, thereby performing priority control of voice data or securing a bandwidth dedicated to voice data.
However, VoIP is basically a service of a best effort type and, therefore, has difficulty in always providing with the best communication quality to a subscriber, even if some technologies to improve the communication quality as described above are introduced.
Specifically, sudden decline of voice quality or sudden interruption of communication would prevent a subscriber from using VoIP communication without anxiety.
Likewise, regarding security measures such as cryptographic processing on voice data, a subscriber would find difficulty in setting/resetting an appropriate security process at the subscriber's request or in confirming security-processing status, thereby being prevented from using VoIP communication without anxiety.
In order to provide a subscriber with high communication quality as possible, it is necessary to provide a service in which the subscriber can observe a communication status in an IP network as the subscriber wishes, can select an appropriate condition of voice communication according to the observed communication status of the IP network as the subscriber requests, and can confirm whether or not the IP network complies with the subscriber's request. However, such technology that enables a subscriber to observe a communication status of voice data in an IP network easily via the subscriber's terminal, such as a telephone or a facsimile machine, has not been developed.